The present invention relates to a flow control system and method for avoiding any congestion at a node connected to a packet switching network to execute communication protocol processing.
Currently, packet switching networks (IP networks) using the Internet protocol (IP) are widely used. In an IP network, end-to-end congestion is avoided between transmitting and receiving nodes. That is, in protocol layer 4 serving as the transport layer of the reference model of OSI (Open Systems Interconnection), congestion is avoided by flow control of TCP (Transmission Control Protocol). TCP is a connection-oriented transfer protocol which transfers data on the basis of a virtual circuit (VC). In the flow control of TCP, a window control scheme is used. When a transmitting node detects packet loss in a network, the cause for it is determined as congestion, and the transmission data amount (congestion window) is reduced to ½. A congestion window is an estimated value of the transfer capability of a network.
For more efficient congestion control, ECN (Explicit Congestion Notification) which explicitly notifies a network of a symptom of congestion has been proposed. In ECN, each packet has congestion indication information. A relay node in a network catches a symptom of congestion (when the transmission buffer occupation amount of the relay node becomes large) and sets the information. A receiving node returns to a transmitting node a packet having such congestion information. With this scheme, the transmitting node reduces the congestion window to ½, as in the case of packet loss.
The operation of TCP is described in Japanese Patent Laid-Open No. 2000-134279 (reference 1). References to be mentioned include “TCP Timeout and Retransmission”, TCP Illustrated Vol. 1 (Addison Wesley, 1994), Chapter 21, pp. 297–322 and Internet Engineering Task Force (IETF), Request for Comments (RFC): 2001/2481, January 1997/January 1999 (references 2 and 3). Reference 1 discloses a method of avoiding unnecessary congestion window reduction when packet loss/rejection has occurred due to not congestion but a line error. Japanese Patent Laid-Open No. 2000-115239 (reference 4) discloses a means for controlling packet retransmission for communication which is managed for each packet and has low management priority while monitoring the congestion state.
An IP network is originally used as a network for data transmission. However, it is recently used for real-time transmission (streaming) of stream data such as voice or image data. The easiest streaming is transmission of stream data encoded at a fixed bit rate. As a transmission protocol suitable for streaming, the Internet Engineering Task Force (IETF), Request for Comments (RFC): 1889, January 1996 (reference 5) is known. However, RTP has no congestion avoiding function. For this reason, if data in an amount beyond the network capacity is to be transmitted, congestion occurs, and packet loss often happens. While congestion is continuing, voice is interrupted, and an image becomes inaccurate. In addition, data transfer using TCP cannot be executed. The reason for this is as follows. Upon detecting congestion, TCP halves the congestion window. However, if congestion continues due to RTP, data that can be transmitted by TCP is exponentially decreased (repeatedly halved) and becomes almost zero.
On the other hand, when TCP is used as a transmission protocol, the transmission bit rate changes in accordance with the state of the network. Hence, the time required until the end of transmission is undefined. For file transfer, only the wait time until transfer is ended becomes long. This does not affect the quality (when all the contents are eventually transferred to the receiving side, no problem occurs in the quality). However, in streaming, reconstruction is sometimes interrupted, resulting in a large degradation in quality (i.e., a time factor is important). That is, in streaming, not only avoiding congestion is necessary but also transmission must be executed while maintaining an expected time continuity (otherwise, reconstruction is stopped halfway, resulting in adverse influence on the quality).
A method of avoiding halfway stop of reconstruction as much as possible while avoiding congestion in streaming has been proposed. In this scheme, a plurality of stream data with different encoding bit rates are prepared and selectively transmitted in accordance with the state of the network. This scheme will be referred to as a stream switching scheme hereinafter. Generally, stream data cannot be simply switched halfway. For this reason, in the stream switching scheme, points (switching points) at which stream data can be switched are prepared for each stream data at a predetermined synchronous time interval (e.g., a 1-sec interval).
For example, Japanese Patent Laid-Open No. 2000-83029 (reference 6) discloses an example of a VOD (video On Demand) system which is a kind of stream switching scheme using an ATM (Asynchronous Transfer Mode) network. In this system, resources are ensured and congestion information is acquired using the function of the ATM network, stream data to be transmitted is switched at a switching point, and the transmission packet rate is updated. However, an IP network itself does not have this function. Since end-to-end congestion is avoided, the technique disclosed in reference 6 cannot be directly applied to the IP network.
In realizing the stream switching scheme on an IP network, the transmission bit rate of TCP is estimated, and stream data is selected and transmitted in accordance with the transmission bit rate. However, in the window control scheme, since a congestion window is controlled, the transmission bit rate is not directly defined. In addition, the delay time in the network, which changes for each packet, directly leads to a variation in transmission bit rate. For this reason, an application measures the amount of data that could be actually transmitted, calculates an average transmission bit rate that is average for a given time, and uses the average transmission bit rate as an estimated value of the transmission bit rate after that.
However, since switching points are present only at a predetermined time interval, continuous reconstruction is not always guaranteed. For example, assume that the average transmission bit rate of the TCP is 1 Mbps, and stream data of 1 Mbps (1,000 kbps) is transmitted in accordance with the transmission bit rate. Assume that the average transmission bit rate suddenly changes to 100 kbps. If stream data (500 kbits) corresponding to 0.5 sec remains until the next switching point, the 500-kbit data is sent at 100 kbps (5 sec is required for transmission). For this reason, an extra delay of 4.5 sec occurs, and reconstruction stops for 4.5 sec.
This problem that reconstruction stops halfway is posed even when an encoding means capable of encoding data in real time is used as an application. Generally, an encoding means has an internal buffer (for, e.g., 0.5 sec at maximum) to smooth the output bit rate. Hence, a code sequence that is already generated and is present in the buffer must be output at the bit rate at the time of encoding. For example, when the transmission bit rate of TCP is 200 kbps, and a 100-kbit encoded code sequence is present in the buffer, the contents of the buffer must be output within 0.5 sec. However, if the transmission bit rate of TCP suddenly changes to 100 kbps, the time required for output changes to 1 sec. When viewed from the receiving node, the delay suddenly increases by 0.5 sec. For this reason, reconstruction stops for 0.5 sec.
As a measure against this problem, a reconstruction delay buffer for a long time (e.g., 10 sec or more) is prepared in the receiving node. If the delay due to the variation in transmission bit rate falls within this range, halfway reconstruction stop can be avoided, and the probability of continuous reconstruction can be increased. Hence, a large delay buffer is generally used in streaming using TCP as a transmission protocol. However, although the probability of continuous reconstruction can be increased, halfway reconstruction stop cannot be completely eliminated. It is difficult to execute a dialogue-type application such as a video phone.
As described above, in the prior arts, if congestion is avoided, reconstruction is stopped halfway. Additionally, to increase the probability of continuous reconstruction, a large reconstruction delay buffer must be inserted to the receiving side. For this reason, high-quality streaming must be abandoned, or the band for streaming must be ensured in advance at the time of network design (a design that eliminates any congestion is necessary).